Jump to content

Calling Speech Server application from a hardware phone via PBXNSIP

zazi

By zazi
in Microsoft Speech Server

 Share

Recommended Posts

zazi Newbie

zazi

Posted
Posted

Hello everybody,

 

I tried hard to get the following setup to work:

I have Speech Server application, where I can successfully connect from softphone (X-Lite client), which is connected to PBXNSIP. Now I like to phone from a hardware telephone. So I sign an account by Sipgate and registered it as a "SIP registration" trunk on my PBXNSIP system. Furthermore, I created an extension with the SIP-ID of Sipgate, that the incomming call has an trunk as "start point". In the Dial plan setup I routed the SIP-ID to the number of my Speech Server application, but everything I got until now is the mailbox (which is now disabled) and that the service is temporarily not available. So I think it should be something with the routing. I thought the configuration should be quite similar to that one of my softphone.

 

Here I have log snippets of both connections:

 

1. from the softphone:

 

[7] 2008/11/23 21:30:05: SIP Rx udp:192.168.1.111:53766:

REGISTER sip:192.168.1.111:7060 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-0f078d3b3f35d84b-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:4321@192.168.1.111:53766;rinstance=3d8ae1d0407da718>

To: "4321"<sip:4321@192.168.1.111:7060>

From: "4321"<sip:4321@192.168.1.111:7060>;tag=3b379f04

Call-ID: OGEzN2JlY2RmNzQ5ZDMwNjhkN2MzYmU4M2FiNmEyMDE.

CSeq: 7 REGISTER

Expires: 3600

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO

User-Agent: X-Lite release 1100l stamp 47546

Authorization: Digest username="4321",realm="192.168.1.111",nonce="7feed25b310955128f9aeb885c8998ea",uri="sip:192.168.1.111:7060",response="ad9353ae5eda61f1dc7de1763e90ff53",algorithm=MD5

Content-Length: 0

 

 

[9] 2008/11/23 21:30:05: Resolve 331: udp 192.168.1.111 53766

[7] 2008/11/23 21:30:05: SIP Tx udp:192.168.1.111:53766:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-0f078d3b3f35d84b-1---d8754z-;rport=53766;received=192.168.1.111

From: "4321" <sip:4321@192.168.1.111:7060>;tag=3b379f04

To: "4321" <sip:4321@192.168.1.111:7060>;tag=75f488e4b9

Call-ID: OGEzN2JlY2RmNzQ5ZDMwNjhkN2MzYmU4M2FiNmEyMDE.

CSeq: 7 REGISTER

Contact: <sip:4321@192.168.1.111:53766;rinstance=3d8ae1d0407da718>;expires=28

Content-Length: 0

 

 

[7] 2008/11/23 21:30:08: SIP Rx udp:192.168.1.111:53766:

PUBLISH sip:4321@192.168.1.111:7060 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-c030d1640b747368-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:4321@192.168.1.111:53766>

To: "4321"<sip:4321@192.168.1.111:7060>

From: "4321"<sip:4321@192.168.1.111:7060>;tag=0d184055

Call-ID: NmRmZGE5MmUxOWUxMmY5ZWE2Mzg2MWJmNjNiYzIxODk.

CSeq: 2 PUBLISH

Expires: 3600

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO

Content-Type: application/pidf+xml

SIP-If-Match: b9g4pa

User-Agent: X-Lite release 1100l stamp 47546

Event: presence

Content-Length: 450

 

<?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity='sip:4321@192.16'><tuple id='t6b2be110'><status><basic>open</basic></status></tuple><dm:person id='t6b2be110'><rpid:activities><rpid:on-the-phone/></rpid:activities><dm:note>On the Phone</dm:note></dm:person></presence>

[9] 2008/11/23 21:30:08: Resolve 332: udp 192.168.1.111 53766

[7] 2008/11/23 21:30:08: SIP Tx udp:192.168.1.111:53766:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-c030d1640b747368-1---d8754z-;rport=53766;received=192.168.1.111

From: "4321" <sip:4321@192.168.1.111:7060>;tag=0d184055

To: "4321" <sip:4321@192.168.1.111:7060>;tag=55447f1982

Call-ID: NmRmZGE5MmUxOWUxMmY5ZWE2Mzg2MWJmNjNiYzIxODk.

CSeq: 2 PUBLISH

SIP-ETag: b9g4pa

Expires: 3600

Content-Length: 0

 

 

[5] 2008/11/23 21:30:08: SIP port accept from 192.168.1.111:1421

[7] 2008/11/23 21:30:10: SIP Rx udp:192.168.1.111:53766:

INVITE sip:0814@192.168.1.111 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-d254ed1b001ce96e-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:4321@192.168.1.111:53766>

To: "PizzaOrder"<sip:0814@192.168.1.111>

From: "4321"<sip:4321@192.168.1.111:7060>;tag=18323e09

Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA.

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO

Content-Type: application/sdp

User-Agent: X-Lite release 1100l stamp 47546

Content-Length: 484

 

v=0

o=- 4 2 IN IP4 192.168.1.111

s=CounterPath X-Lite 3.0

c=IN IP4 192.168.1.111

t=0 0

m=audio 53768 RTP/AVP 107 119 100 106 6 0 97 105 98 8 102 3 5 101

a=alt:1 1 : IF9wAIIz Urysamk7 192.168.1.111 53768

a=fmtp:101 0-15

a=rtpmap:107 BV32/16000

a=rtpmap:119 BV32-FEC/16000

a=rtpmap:100 SPEEX/16000

a=rtpmap:106 SPEEX-FEC/16000

a=rtpmap:97 SPEEX/8000

a=rtpmap:105 SPEEX-FEC/8000

a=rtpmap:98 iLBC/8000

a=rtpmap:102 L16/16000

a=rtpmap:101 telephone-event/8000

a=sendrecv

 

[9] 2008/11/23 21:30:10: UDP: Opening socket on port 51420

[9] 2008/11/23 21:30:10: UDP: Opening socket on port 51421

[8] 2008/11/23 21:30:10: Could not find a trunk (3 trunks)

[8] 2008/11/23 21:30:10: Using outbound proxy sip:192.168.1.111:53766;transport=udp because UDP packet source did not match the via header

[9] 2008/11/23 21:30:10: Resolve 333: udp 192.168.1.111 53766

[7] 2008/11/23 21:30:10: SIP Tx udp:192.168.1.111:53766:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-d254ed1b001ce96e-1---d8754z-;rport=53766;received=192.168.1.111

From: "4321" <sip:4321@192.168.1.111:7060>;tag=18323e09

To: "PizzaOrder" <sip:0814@192.168.1.111>;tag=ffc0e8c2c1

Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA.

CSeq: 1 INVITE

Content-Length: 0

 

 

[9] 2008/11/23 21:30:10: Resolve 334: udp 192.168.1.111 53766

[7] 2008/11/23 21:30:10: SIP Tx udp:192.168.1.111:53766:

SIP/2.0 401 Authentication Required

Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-d254ed1b001ce96e-1---d8754z-;rport=53766;received=192.168.1.111

From: "4321" <sip:4321@192.168.1.111:7060>;tag=18323e09

To: "PizzaOrder" <sip:0814@192.168.1.111>;tag=ffc0e8c2c1

Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA.

CSeq: 1 INVITE

User-Agent: pbxnsip-PBX/3.0.1.3023

WWW-Authenticate: Digest realm="192.168.1.111",nonce="34e0ecd53dc31bfd7464f7b7a7942fd2",domain="sip:0814@192.168.1.111",algorithm=MD5

Content-Length: 0

 

 

[7] 2008/11/23 21:30:10: SIP Rx udp:192.168.1.111:53766:

ACK sip:0814@192.168.1.111 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-d254ed1b001ce96e-1---d8754z-;rport

To: "PizzaOrder" <sip:0814@192.168.1.111>;tag=ffc0e8c2c1

From: "4321"<sip:4321@192.168.1.111:7060>;tag=18323e09

Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA.

CSeq: 1 ACK

Content-Length: 0

 

 

[7] 2008/11/23 21:30:10: SIP Rx udp:192.168.1.111:53766:

INVITE sip:0814@192.168.1.111 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-b2608d762672484e-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:4321@192.168.1.111:53766>

To: "PizzaOrder"<sip:0814@192.168.1.111>

From: "4321"<sip:4321@192.168.1.111:7060>;tag=18323e09

Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA.

CSeq: 2 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO

Content-Type: application/sdp

User-Agent: X-Lite release 1100l stamp 47546

Authorization: Digest username="4321",realm="192.168.1.111",nonce="34e0ecd53dc31bfd7464f7b7a7942fd2",uri="sip:0814@192.168.1.111",response="c02ce56f89a56e0498b5d0832739958a",algorithm=MD5

Content-Length: 484

 

v=0

o=- 4 2 IN IP4 192.168.1.111

s=CounterPath X-Lite 3.0

c=IN IP4 192.168.1.111

t=0 0

m=audio 53768 RTP/AVP 107 119 100 106 6 0 97 105 98 8 102 3 5 101

a=alt:1 1 : IF9wAIIz Urysamk7 192.168.1.111 53768

a=fmtp:101 0-15

a=rtpmap:107 BV32/16000

a=rtpmap:119 BV32-FEC/16000

a=rtpmap:100 SPEEX/16000

a=rtpmap:106 SPEEX-FEC/16000

a=rtpmap:97 SPEEX/8000

a=rtpmap:105 SPEEX-FEC/8000

a=rtpmap:98 iLBC/8000

a=rtpmap:102 L16/16000

a=rtpmap:101 telephone-event/8000

a=sendrecv

 

[8] 2008/11/23 21:30:10: Tagging request with existing tag

[6] 2008/11/23 21:30:10: Sending RTP for Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA.#ffc0e8c2c1 to 192.168.1.111:53768

[9] 2008/11/23 21:30:10: Resolve 335: udp 192.168.1.111 53766

[7] 2008/11/23 21:30:10: SIP Tx udp:192.168.1.111:53766:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-b2608d762672484e-1---d8754z-;rport=53766;received=192.168.1.111

From: "4321" <sip:4321@192.168.1.111:7060>;tag=18323e09

To: "PizzaOrder" <sip:0814@192.168.1.111>;tag=ffc0e8c2c1

Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA.

CSeq: 2 INVITE

Content-Length: 0

 

 

[9] 2008/11/23 21:30:10: Dialplan: Evaluating !^(4321)@.*!sip:0814@\r;user=phone!i against 0814@192.168.1.111

[9] 2008/11/23 21:30:10: Dialplan: Evaluating !^(0814)@.*!sip:\1@\r;user=phone!i against 0814@192.168.1.111

[5] 2008/11/23 21:30:10: Dialplan PizzaOrder: Match 0814@192.168.1.111 to <sip:0814@192.168.1.111;user=phone> on trunk MSSpeechServer

[5] 2008/11/23 21:30:10: Charge user 4321 for redirecting calls

[8] 2008/11/23 21:30:10: Play audio_moh/noise.wav

[9] 2008/11/23 21:30:10: UDP: Opening socket on port 59724

[9] 2008/11/23 21:30:10: UDP: Opening socket on port 59725

[9] 2008/11/23 21:30:10: Resolve 336: url sip:192.168.1.111:15060;transport=tcp

[9] 2008/11/23 21:30:10: Resolve 336: a tcp 192.168.1.111 15060

[9] 2008/11/23 21:30:10: Resolve 336: tcp 192.168.1.111 15060

[7] 2008/11/23 21:30:10: SIP Tx tcp:192.168.1.111:15060:

INVITE sip:0814@192.168.1.111;user=phone SIP/2.0

Via: SIP/2.0/TCP 127.0.0.1:1423;branch=z9hG4bK-a79b4caf22387f8f9ab1bc30cb7925c7;rport

From: "zazi" <sip:100@192.168.1.111;user=phone>;tag=10941

To: <sip:0814@192.168.1.111;user=phone>

Call-ID: 8493d05b@pbx

CSeq: 5041 INVITE

Max-Forwards: 70

Contact: <sip:100@127.0.0.1:1423;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.1.3023

Content-Type: application/sdp

Content-Length: 284

 

v=0

o=- 63976 63976 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 59724 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2008/11/23 21:30:10: SIP Rx tcp:192.168.1.111:15060:

SIP/2.0 100 Trying

FROM: "zazi"<sip:100@192.168.1.111;user=phone>;tag=10941

TO: <sip:0814@192.168.1.111;user=phone>

CSEQ: 5041 INVITE

CALL-ID: 8493d05b@pbx

VIA: SIP/2.0/TCP 127.0.0.1:1423;branch=z9hG4bK-a79b4caf22387f8f9ab1bc30cb7925c7;rport

CONTENT-LENGTH: 0

 

 

[9] 2008/11/23 21:30:10: Resolve 337: udp 192.168.1.111 53766

[7] 2008/11/23 21:30:10: SIP Tx udp:192.168.1.111:53766:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-b2608d762672484e-1---d8754z-;rport=53766;received=192.168.1.111

From: "4321" <sip:4321@192.168.1.111:7060>;tag=18323e09

To: "PizzaOrder" <sip:0814@192.168.1.111>;tag=ffc0e8c2c1

Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA.

CSeq: 2 INVITE

Contact: <sip:4321@127.0.0.1:7060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.1.3023

Content-Type: application/sdp

Content-Length: 233

 

v=0

o=- 15820 15820 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 51420 RTP/AVP 0 8 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2008/11/23 21:30:10: SIP Rx tcp:192.168.1.111:15060:

SIP/2.0 302 Moved Temporarily

FROM: "zazi"<sip:100@192.168.1.111;user=phone>;tag=10941

TO: <sip:0814@192.168.1.111;user=phone>;tag=3899c73dd5

CSEQ: 5041 INVITE

CALL-ID: 8493d05b@pbx

VIA: SIP/2.0/TCP 127.0.0.1:1423;branch=z9hG4bK-a79b4caf22387f8f9ab1bc30cb7925c7;rport

CONTACT: <sip:0814@192.168.1.111:6060;user=phone;transport=Tcp;maddr=192.168.1.111;x-mss-call-id=8493d05b%40pbx>

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[7] 2008/11/23 21:30:10: Call 8493d05b@pbx#10941: Clear last INVITE

[9] 2008/11/23 21:30:10: Resolve 338: url sip:192.168.1.111:15060;transport=tcp

[9] 2008/11/23 21:30:10: Resolve 338: a tcp 192.168.1.111 15060

[9] 2008/11/23 21:30:10: Resolve 338: tcp 192.168.1.111 15060

[7] 2008/11/23 21:30:10: SIP Tx tcp:192.168.1.111:15060:

ACK sip:0814@192.168.1.111;user=phone SIP/2.0

Via: SIP/2.0/TCP 127.0.0.1:1423;branch=z9hG4bK-a79b4caf22387f8f9ab1bc30cb7925c7;rport

From: "zazi" <sip:100@192.168.1.111;user=phone>;tag=10941

To: <sip:0814@192.168.1.111;user=phone>;tag=3899c73dd5

Call-ID: 8493d05b@pbx

CSeq: 5041 ACK

Max-Forwards: 70

Contact: <sip:100@127.0.0.1:1423;transport=tcp>

Content-Length: 0

 

 

[5] 2008/11/23 21:30:10: Redirecting call

[9] 2008/11/23 21:30:10: Resolve 339: aaaa tcp 192.168.1.111 6060

[9] 2008/11/23 21:30:10: Resolve 339: a tcp 192.168.1.111 6060

[9] 2008/11/23 21:30:10: Resolve 339: tcp 192.168.1.111 6060

[7] 2008/11/23 21:30:10: SIP Tx tcp:192.168.1.111:6060:

INVITE sip:0814@192.168.1.111:6060;user=phone;transport=Tcp;maddr=192.168.1.111;x-mss-call-id=8493d05b%40pbx SIP/2.0

Via: SIP/2.0/TCP 127.0.0.1:1425;branch=z9hG4bK-0f4d1da0b6d4b796600a6621b56f3977;rport

From: "zazi" <sip:100@192.168.1.111;user=phone>;tag=10941

To: <sip:0814@192.168.1.111;user=phone>

Call-ID: 8493d05b@pbx

CSeq: 5042 INVITE

Max-Forwards: 70

Contact: <sip:100@127.0.0.1:1425;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.1.3023

Content-Type: application/sdp

Content-Length: 284

 

v=0

o=- 63976 63976 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 59724 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[6] 2008/11/23 21:30:10: Sending RTP for Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA.#ffc0e8c2c1 to 127.0.0.1:53768

[7] 2008/11/23 21:30:10: SIP Rx tcp:192.168.1.111:6060:

SIP/2.0 100 Trying

FROM: "zazi"<sip:100@192.168.1.111;user=phone>;tag=10941

TO: <sip:0814@192.168.1.111;user=phone>

CSEQ: 5042 INVITE

CALL-ID: 8493d05b@pbx

VIA: SIP/2.0/TCP 127.0.0.1:1425;branch=z9hG4bK-0f4d1da0b6d4b796600a6621b56f3977;rport

CONTENT-LENGTH: 0

 

 

[7] 2008/11/23 21:30:10: SIP Rx tcp:192.168.1.111:6060:

SIP/2.0 180 Ringing

FROM: "zazi"<sip:100@192.168.1.111;user=phone>;tag=10941

TO: <sip:0814@192.168.1.111;user=phone>;epid=3E41C36034;tag=392e214f18

CSEQ: 5042 INVITE

CALL-ID: 8493d05b@pbx

VIA: SIP/2.0/TCP 127.0.0.1:1425;branch=z9hG4bK-0f4d1da0b6d4b796600a6621b56f3977;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[8] 2008/11/23 21:30:10: Play audio_en/ringback.wav

[6] 2008/11/23 21:30:10: Sending RTP for 8493d05b@pbx#10941 to 192.168.1.111:13440

[7] 2008/11/23 21:30:10: SIP Rx tcp:192.168.1.111:6060:

SIP/2.0 200 OK

FROM: "zazi"<sip:100@192.168.1.111;user=phone>;tag=10941

TO: <sip:0814@192.168.1.111;user=phone>;epid=3E41C36034;tag=392e214f18

CSEQ: 5042 INVITE

CALL-ID: 8493d05b@pbx

VIA: SIP/2.0/TCP 127.0.0.1:1425;branch=z9hG4bK-0f4d1da0b6d4b796600a6621b56f3977;rport

CONTACT: <sip:LEPPI01:6060;transport=Tcp;maddr=192.168.1.111>;automata

CONTENT-LENGTH: 196

CONTENT-TYPE: application/sdp

ALLOW: UPDATE

SERVER: RTCC/3.0.0.0

ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

 

v=0

o=- 0 0 IN IP4 192.168.1.111

s=Microsoft Speech Server session

c=IN IP4 192.168.1.111

t=0 0

m=audio 13440 RTP/AVP 0 8 101

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

 

[7] 2008/11/23 21:30:10: Call 8493d05b@pbx#10941: Clear last INVITE

[7] 2008/11/23 21:30:10: Set packet length to 20

[9] 2008/11/23 21:30:10: Resolve 340: aaaa tcp 192.168.1.111 6060

[9] 2008/11/23 21:30:10: Resolve 340: a tcp 192.168.1.111 6060

[9] 2008/11/23 21:30:10: Resolve 340: tcp 192.168.1.111 6060

[7] 2008/11/23 21:30:10: SIP Tx tcp:192.168.1.111:6060:

ACK sip:LEPPI01:6060;transport=Tcp;maddr=192.168.1.111 SIP/2.0

Via: SIP/2.0/TCP 127.0.0.1:1425;branch=z9hG4bK-fee5e5c4eb71384086a295ec5034ac76;rport

From: "zazi" <sip:100@192.168.1.111;user=phone>;tag=10941

To: <sip:0814@192.168.1.111;user=phone>;tag=392e214f18

Call-ID: 8493d05b@pbx

CSeq: 5042 ACK

Max-Forwards: 70

Contact: <sip:100@127.0.0.1:1425;transport=tcp>

Content-Length: 0

 

 

[7] 2008/11/23 21:30:10: Determine pass-through mode after receiving response

[9] 2008/11/23 21:30:10: Resolve 341: udp 192.168.1.111 53766

[7] 2008/11/23 21:30:10: SIP Tx udp:192.168.1.111:53766:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-b2608d762672484e-1---d8754z-;rport=53766;received=192.168.1.111

From: "4321" <sip:4321@192.168.1.111:7060>;tag=18323e09

To: "PizzaOrder" <sip:0814@192.168.1.111>;tag=ffc0e8c2c1

Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA.

CSeq: 2 INVITE

Contact: <sip:4321@127.0.0.1:7060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.1.3023

Content-Type: application/sdp

Content-Length: 233

 

v=0

o=- 15820 15820 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 51420 RTP/AVP 0 8 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2008/11/23 21:30:10: SIP Rx udp:127.0.0.1:53766:

ACK sip:4321@127.0.0.1:7060 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-f90fa7068f2f5109-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:4321@192.168.1.111:53766>

To: "PizzaOrder"<sip:0814@192.168.1.111>;tag=ffc0e8c2c1

From: "4321"<sip:4321@192.168.1.111:7060>;tag=18323e09

Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA.

CSeq: 2 ACK

User-Agent: X-Lite release 1100l stamp 47546

Authorization: Digest username="4321",realm="192.168.1.111",nonce="34e0ecd53dc31bfd7464f7b7a7942fd2",uri="sip:0814@192.168.1.111",response="c02ce56f89a56e0498b5d0832739958a",algorithm=MD5

Content-Length: 0

 

 

[7] 2008/11/23 21:30:10: 8493d05b@pbx#10941: RTP pass-through mode

[7] 2008/11/23 21:30:10: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA.#ffc0e8c2c1: RTP pass-through mode

[7] 2008/11/23 21:30:13: SIP Rx udp:127.0.0.1:53766:

BYE sip:4321@127.0.0.1:7060 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-d6017e10377e7a0d-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:4321@192.168.1.111:53766>

To: "PizzaOrder"<sip:0814@192.168.1.111>;tag=ffc0e8c2c1

From: "4321"<sip:4321@192.168.1.111:7060>;tag=18323e09

Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA.

CSeq: 3 BYE

User-Agent: X-Lite release 1100l stamp 47546

Authorization: Digest username="4321",realm="192.168.1.111",nonce="34e0ecd53dc31bfd7464f7b7a7942fd2",uri="sip:4321@127.0.0.1:7060",response="044d4a36644d99fadc4e61330d41253f",algorithm=MD5

Reason: SIP;description="User Hung Up"

Content-Length: 0

 

 

[9] 2008/11/23 21:30:13: Resolve 342: aaaa udp 127.0.0.1 53766

[9] 2008/11/23 21:30:13: Resolve 342: a udp 127.0.0.1 53766

[9] 2008/11/23 21:30:13: Resolve 342: udp 127.0.0.1 53766

[7] 2008/11/23 21:30:13: SIP Tx udp:127.0.0.1:53766:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-d6017e10377e7a0d-1---d8754z-;rport=53766

From: "4321" <sip:4321@192.168.1.111:7060>;tag=18323e09

To: "PizzaOrder" <sip:0814@192.168.1.111>;tag=ffc0e8c2c1

Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA.

CSeq: 3 BYE

Contact: <sip:4321@127.0.0.1:7060>

User-Agent: pbxnsip-PBX/3.0.1.3023

RTP-RxStat: Dur=3,Pkt=146,Oct=25112,Underun=0

RTP-TxStat: Dur=3,Pkt=147,Oct=25284

Content-Length: 0

 

 

[7] 2008/11/23 21:30:13: 8493d05b@pbx#10941: Media-aware pass-through mode

[7] 2008/11/23 21:30:13: Other Ports: 1

[7] 2008/11/23 21:30:13: Call Port: 8493d05b@pbx#10941

[7] 2008/11/23 21:30:13: SIP Rx udp:192.168.1.111:53766:

PUBLISH sip:4321@192.168.1.111:7060 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-114af4181f09301a-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:4321@192.168.1.111:53766>

To: "4321"<sip:4321@192.168.1.111:7060>

From: "4321"<sip:4321@192.168.1.111:7060>;tag=0d184055

Call-ID: NmRmZGE5MmUxOWUxMmY5ZWE2Mzg2MWJmNjNiYzIxODk.

CSeq: 3 PUBLISH

Expires: 3600

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO

Content-Type: application/pidf+xml

SIP-If-Match: b9g4pa

User-Agent: X-Lite release 1100l stamp 47546

Event: presence

Content-Length: 414

 

<?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity='sip:4321@192.16'><tuple id='t6b2be110'><status><basic>open</basic></status></tuple><dm:person id='t6b2be110'><rpid:activities><rpid:unknown/></rpid:activities></dm:person></presence>

[9] 2008/11/23 21:30:13: Resolve 343: udp 192.168.1.111 53766

[7] 2008/11/23 21:30:13: SIP Tx udp:192.168.1.111:53766:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-114af4181f09301a-1---d8754z-;rport=53766;received=192.168.1.111

From: "4321" <sip:4321@192.168.1.111:7060>;tag=0d184055

To: "4321" <sip:4321@192.168.1.111:7060>;tag=55447f1982

Call-ID: NmRmZGE5MmUxOWUxMmY5ZWE2Mzg2MWJmNjNiYzIxODk.

CSeq: 3 PUBLISH

SIP-ETag: b9g4pa

Expires: 3600

Content-Length: 0

 

 

[9] 2008/11/23 21:30:13: Resolve 344: aaaa tcp 192.168.1.111 6060

[9] 2008/11/23 21:30:13: Resolve 344: a tcp 192.168.1.111 6060

[9] 2008/11/23 21:30:13: Resolve 344: tcp 192.168.1.111 6060

[7] 2008/11/23 21:30:13: SIP Tx tcp:192.168.1.111:6060:

BYE sip:LEPPI01:6060;transport=Tcp;maddr=192.168.1.111 SIP/2.0

Via: SIP/2.0/TCP 127.0.0.1:1425;branch=z9hG4bK-d0cd32ed84b1bf136da63e679df6f801;rport

From: "zazi" <sip:100@192.168.1.111;user=phone>;tag=10941

To: <sip:0814@192.168.1.111;user=phone>;tag=392e214f18

Call-ID: 8493d05b@pbx

CSeq: 5043 BYE

Max-Forwards: 70

Contact: <sip:100@127.0.0.1:1425;transport=tcp>

RTP-RxStat: Dur=3,Pkt=133,Oct=22876,Underun=0

RTP-TxStat: Dur=3,Pkt=129,Oct=22188

Content-Length: 0

 

 

[7] 2008/11/23 21:30:13: SIP Rx tcp:192.168.1.111:6060:

SIP/2.0 200 OK

FROM: "zazi"<sip:100@192.168.1.111;user=phone>;tag=10941

TO: <sip:0814@192.168.1.111;user=phone>;tag=392e214f18;epid=3E41C36034

CSEQ: 5043 BYE

CALL-ID: 8493d05b@pbx

VIA: SIP/2.0/TCP 127.0.0.1:1425;branch=z9hG4bK-d0cd32ed84b1bf136da63e679df6f801;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[7] 2008/11/23 21:30:13: Call 8493d05b@pbx#10941: Clear last request

[5] 2008/11/23 21:30:13: BYE Response: Terminate 8493d05b@pbx

 

 

==================================================================

 

53766 is the port where the softphone client is listening

7060 is the tcp port of my PBXNSIP system

6060 is the port of my Speech Server application

100 is the Trunk ANI of the Speech Server SIP gateway

4321 is the number/name of my softphone client, which is an extension account at my PBXNSIP system

PizzaOrder is the name of my dial plan (because it should connect to the PizzaOrder tutorial application from MS Speech Server)

 

==================================================================

 

2. from my hardphone:

 

[7] 2008/11/23 20:41:00: SIP Rx udp:217.10.79.9:5060:

INVITE sip:[my Sipgate SIP-ID]@192.168.2.100:7060;transport=udp;line=a87ff679 SIP/2.0

Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d>

Record-Route: <sip:172.20.40.2;lr=on>

Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d>

Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKf4c8.91cbc5e.0

Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bKf4c8.91cbc5e.0

Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK54fb519d

Via: SIP/2.0/UDP 217.10.67.5:5060;branch=z9hG4bK54fb519d;rport=5060

From: "[my real hardphone number]" <sip:[my real hardphone number]@sipgate.de>;tag=as1a76684d

To: <sip:[my Sipgate phone number with country prefix]@sipgate.de>

Contact: <sip:[my real hardphone number]@217.10.67.5>

Call-ID: 7992bc44273786571088032e273db69c@sipgate.de

CSeq: 102 INVITE

Max-Forwards: 67

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Type: application/sdp

Content-Length: 408

 

v=0

o=root 24764 24764 IN IP4 217.10.67.5

s=session

c=IN IP4 217.10.67.5

t=0 0

m=audio 11354 RTP/AVP 8 0 3 97 18 112 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:97 iLBC/8000

a=fmtp:97 mode=30

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:112 G726-32/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

 

[9] 2008/11/23 20:41:00: UDP: Opening socket on port 54212

[9] 2008/11/23 20:41:00: UDP: Opening socket on port 54213

[5] 2008/11/23 20:41:00: Identify trunk (line match) 4

[9] 2008/11/23 20:41:00: Resolve 221: aaaa udp 217.10.79.9 5060

[9] 2008/11/23 20:41:00: Resolve 221: a udp 217.10.79.9 5060

[9] 2008/11/23 20:41:00: Resolve 221: udp 217.10.79.9 5060

[7] 2008/11/23 20:41:00: SIP Tx udp:217.10.79.9:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKf4c8.91cbc5e.0

Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bKf4c8.91cbc5e.0

Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK54fb519d

Via: SIP/2.0/UDP 217.10.67.5:5060;branch=z9hG4bK54fb519d;rport=5060

Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d>

Record-Route: <sip:172.20.40.2;lr=on>

Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d>

From: "[my real hardphone number]" <sip:[my real hardphone number]@sipgate.de>;tag=as1a76684d

To: <sip:[my Sipgate phone number with country prefix]@sipgate.de>;tag=0d0d85e1a1

Call-ID: 7992bc44273786571088032e273db69c@sipgate.de

CSeq: 102 INVITE

Content-Length: 0

 

 

[7] 2008/11/23 20:41:00: Set packet length to 20

[6] 2008/11/23 20:41:00: Sending RTP for 7992bc44273786571088032e273db69c@sipgate.de#0d0d85e1a1 to 217.10.67.5:11354

[5] 2008/11/23 20:41:00: Trunk Sipgate sends call to [my Sipgate SIP-ID] in domain pbx.company.com

[7] 2008/11/23 20:41:00: Attendant: Calling extension [my Sipgate SIP-ID]

[5] 2008/11/23 20:41:00: Attendant: Redirect to

[9] 2008/11/23 20:41:00: Dialplan: Evaluating !^(4321)@.*!sip:0814@\r;user=phone!i against sip:pbx.company.com@pbx.company.com

[9] 2008/11/23 20:41:00: Dialplan: Evaluating !^(0814)@.*!sip:\1@\r;user=phone!i against sip:pbx.company.com@pbx.company.com

[9] 2008/11/23 20:41:00: Dialplan: Evaluating !^9181([0-9]*)@.*!sip:0814@\r;user=phone!i against sip:pbx.company.com@pbx.company.com

[9] 2008/11/23 20:41:00: Dialplan: Evaluating !^0351([0-9]*)@.*!sip:0814@\r;user=phone!i against sip:pbx.company.com@pbx.company.com

[9] 2008/11/23 20:41:00: Dialplan: Evaluating !^9181([0-9]*)@.*!sip:0814@\r;user=phone!i against sip:pbx.company.com@pbx.company.com

[9] 2008/11/23 20:41:00: Dialplan: Evaluating !^0049351([0-9]*)@.*!sip:0814@\r;user=phone!i against sip:pbx.company.com@pbx.company.com

[7] 2008/11/23 20:41:00: Set packet length to 20

[9] 2008/11/23 20:41:00: Resolve 222: aaaa udp 217.10.79.9 5060

[9] 2008/11/23 20:41:00: Resolve 222: a udp 217.10.79.9 5060

[9] 2008/11/23 20:41:00: Resolve 222: udp 217.10.79.9 5060

[7] 2008/11/23 20:41:00: SIP Tx udp:217.10.79.9:5060:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKf4c8.91cbc5e.0

Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bKf4c8.91cbc5e.0

Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK54fb519d

Via: SIP/2.0/UDP 217.10.67.5:5060;branch=z9hG4bK54fb519d;rport=5060

Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d>

Record-Route: <sip:172.20.40.2;lr=on>

Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d>

From: "[my real hardphone number]" <sip:[my real hardphone number]@sipgate.de>;tag=as1a76684d

To: <sip:[my Sipgate phone number with country prefix]@sipgate.de>;tag=0d0d85e1a1

Call-ID: 7992bc44273786571088032e273db69c@sipgate.de

CSeq: 102 INVITE

Contact: <sip:[my Sipgate SIP-ID]@192.168.1.111:7060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.1.3023

Content-Length: 0

 

 

[9] 2008/11/23 20:41:00: Resolve 223: aaaa udp 217.10.79.9 5060

[9] 2008/11/23 20:41:00: Resolve 223: a udp 217.10.79.9 5060

[9] 2008/11/23 20:41:00: Resolve 223: udp 217.10.79.9 5060

[7] 2008/11/23 20:41:00: SIP Tx udp:217.10.79.9:5060:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKf4c8.91cbc5e.0

Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bKf4c8.91cbc5e.0

Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK54fb519d

Via: SIP/2.0/UDP 217.10.67.5:5060;branch=z9hG4bK54fb519d;rport=5060

Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d>

Record-Route: <sip:172.20.40.2;lr=on>

Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d>

From: "[my real hardphone number]" <sip:[my real hardphone number]@sipgate.de>;tag=as1a76684d

To: <sip:[my Sipgate phone number with country prefix]@sipgate.de>;tag=0d0d85e1a1

Call-ID: 7992bc44273786571088032e273db69c@sipgate.de

CSeq: 102 INVITE

Contact: <sip:[my Sipgate SIP-ID]@192.168.1.111:7060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.1.3023

Content-Length: 0

 

 

[7] 2008/11/23 20:41:00: SIP Tr udp:217.10.79.9:5060:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKf4c8.91cbc5e.0

Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bKf4c8.91cbc5e.0

Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK54fb519d

Via: SIP/2.0/UDP 217.10.67.5:5060;branch=z9hG4bK54fb519d;rport=5060

Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d>

Record-Route: <sip:172.20.40.2;lr=on>

Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d>

From: "[my real hardphone number]" <sip:[my real hardphone number]@sipgate.de>;tag=as1a76684d

To: <sip:[my Sipgate phone number with country prefix]@sipgate.de>;tag=0d0d85e1a1

Call-ID: 7992bc44273786571088032e273db69c@sipgate.de

CSeq: 102 INVITE

Contact: <sip:[my Sipgate SIP-ID]@192.168.1.111:7060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.1.3023

Content-Length: 0

 

============================================================

 

At last, I thought it has to something with the country code, because the Sipgate phone number comes along with the full country prefix, but I also add that to my dial plan. As you can see in the last log above I tried different routing, without any positive result. A bad side effect is after one call via the Sipgate SIP gateway the registration get and I have to reboot my system.

 

 

Thanks a lot for any help.

 

Cheers zazi

Link to comment
Share on other sites
zazi Newbie

zazi

Posted
  • Author
Posted

Hi guys,

 

I found the solution. I forwarded the call to 0814@[iP of my SpeechServer];transport=tcp. This construct isn't good for PBXNSIP in that field (as you maybe can see the: after "redirect to" in the log above is nothing before the system begins to check agains the dial plan). So I changed it just to 0814 (the number of my Speech Server application, but it this it not really relevant, because you have to add this again in your dial plan). Now it like to validate against +49814 (it adds automatically the country prefix). So you have to add it again to the dial plan and for the replacement I took now 0814 and it connects to my Speech Server application. Unfortunatelly it do not reconizes my voice input - maybe my telephone is so bad or something else.

 

Hope that will maybe help other ones.

 

Cheers zazi

Link to comment
Share on other sites
  • 2 months later...
Jan Boguslawski Newbie

Jan Boguslawski

Posted
Posted

Hi zazi, :(

 

what kind of phone are you using? What about an outgoing call from Speech Server to this phone?

 

Regards,

 

Jan

Link to comment
Share on other sites
  • 1 month later...
zazi Newbie

zazi

Posted
  • Author
Posted
Hi zazi, :)

 

what kind of phone are you using? What about an outgoing call from Speech Server to this phone?

 

Regards,

 

Jan

 

Hi Jan,

 

through the Sipgate VoIP provider we connected our system to a real telephone number so you can use every kind of telephone or landline. That was a prior design goal of our application. Of course, you can also use a softphone client (in ways: 1. you call directly the delegated telephone number 2. you register the softphone client directly at the ip-pbx. The outgoing call should go over the Message Queuing system, which is related to the hosted speech application. And then know way over the system: the ip-pbx delegated it to the sipgate phone number and the initiate the real call to a real number. Unfortunatelly, this is currently not implemented in our system. We are just sending SMS outside, but therefore we use directly the Webservice from Sipgate.

 

Cheers zazi

 

PS: sorry for the late reply, I've disabled the email notification ;)

Link to comment
Share on other sites
  • 4 weeks later...
zazi Newbie

zazi

Posted
  • Author
Posted

Hello again,

 

now we got also the outbound call running. Here is a short description how to it (please ask for a detail one, if necessary):

 

1. set the proxy at the makeCall activity to your SIP peer of your pbx(nsip)

2. create an account with the specified name of your calling party (from the makeCall activity)

3. route the outbound call to your outbound proxy (I used here my sipgate account)

- it runs here as "Outbound Proxy"

- maybe also important to set up the port of the outbound server

 

Cheers zazi

 

PS: I'm still running version 3.0.1.3023 (because I've read about the problems with P-Asserted-Identify)

Link to comment
Share on other sites
Vodia PBX Grand Master

Vodia PBX

Posted
Posted
PS: I'm still running version 3.0.1.3023 (because I've read about the problems with P-Asserted-Identify)

 

What about inband DTMF? Is that an issue for you? We had a lot of issues recently and added inband to out-of-band transcoding to get that working....

Link to comment
Share on other sites
zazi Newbie

zazi

Posted
  • Author
Posted

Hi,

 

I don't know really what do you mean with "inband DTMF" (it is just my first VUI during my studies). If it is related to the usage of DTMF, then my experience is that DTMF works for inbound calls. For outbound calls it is currently not implemented (so can't test it yet), but we've planned to link the main part of the workflow of our inbound VUI to our outbound VUI.

Generally, our VUI is designed for natural language queries/ communication. So DTMF is just a helping component in the case that the user isn't able to do a natural language query that matches.

 

Cheers zazi

Link to comment
Share on other sites

Join the conversation

You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.

Guest
Reply to this topic...

×   Pasted as rich text.   Paste as plain text instead

  Only 75 emoji are allowed.

×   Your link has been automatically embedded.   Display as a link instead

×   Your previous content has been restored.   Clear editor

×   You cannot paste images directly. Upload or insert images from URL.

×
 Share
Go to topic listing


×
×
  • Create New...